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• Subtitling Sierra VMD Files

  1 June 2016, by Multimedia Mike — Game Hacking

  I was contacted by a game translation hobbyist from Spain (henceforth known as The Translator). He had set his sights on Sierra’s 7-CD Phantasmagoria. This mammoth game was driven by a lot of FMV files and animations that have speech. These require language translation in the form of video subtitling. He’s lucky that he found possibly the one person on the whole internet who has just the right combination of skill, time, and interest to pull this off. And why would I care about helping? I guess I share a certain camaraderie with game hackers. Don’t act so surprised. You know what kind of stuff I like to work on.

  The FMV format used in this game is VMD, which makes an appearance in numerous Sierra titles. FFmpeg already supports decoding this format. FFmpeg also supports subtitling video. So, ideally, all that’s necessary to support this goal is to add a muxer for the VMD format which can encode raw video and audio, which the format supports. Implement video compression as extra credit.

  The pipeline that I envisioned looks like this:

  
  

  

  VMD Subtitling Process

  
  

  “Trivial!” I surmised. I just never learn, do I?

  The Plan
  So here’s my initial pitch, outlining the work I estimated that I would need to do towards the stated goal:

  1. Create a new file muxer that produces a syntactically valid VMD file with bogus video and audio data. Make sure it works with both FFmpeg’s playback system as well as the proper Phantasmagoria engine.
  2. Create a new video encoder that essentially operates in pass-through mode while correctly building a palette.
  3. Create a new basic encoder for the video frames.

  A big unknown for me was exactly how subtitle handling operates in FFmpeg. Thanks to this project, I now know. I was concerned because I was pretty sure that font rendering entails anti-aliasing which bodes poorly for keeping the palette count under 256 unique colors.

  Computer Science Puzzle
  When pondering how to process the palette, I was excited for the opportunity to exercise actual computer science. FFmpeg converts frames from paletted frames to full RGB frames. Then it needs to convert them back to paletted frames. I had a vague recollection of solving this problem once before when I was experimenting with a new paletted video codec. I seem to recall that I did the palette conversion in a very naive manner. I just used a static 256-element array and processed each RGB pixel of the frame, seeing if the value already occurred in the table (O(n) lookup) and adding it otherwise.
  

  There are more efficient algorithms, however, such as hash tables and trees. Somewhere along the line, FFmpeg helpfully acquired a rarely-used tree data structure, which was perfect for this project.

  So I was pretty pleased with this optimization. Too bad this wouldn’t survive to the end of the effort.

  Another palette-related challenge was the fact that a group of pictures would be accumulating a new palette but that palette needed to be recorded before the group. Thus, the muxer needed to have extra logic to rewind the file when the video encoder transmitted a palette change.

  Video Compression
  VMD has a few methods in its compression toolbox. It can use interframe differencing, it has some RLE, or it can code a frame raw. It can also use a custom LZ-like format on top of these. For early prototypes, I elected to leave each frame coded raw. After the concept was proved, I implemented the frame differencing.

  
  

  

  
  Top frame compared with the middle frame yields the bottom frame: red pixels indicate changes
  

  Encoding only those red dots in between vast runs of unchanged pixels yielded a vast measurable improvement. The next step was to try wiring up FFmpeg’s existing LZ compression facilities to the encoder. This turned out to be implausible since VMD’s LZ variant has nothing to do with anything FFmpeg already provides. Fortunately, the LZ piece is not absolutely required and the frame differencing + RLE provides plenty of compression.

  Subtitling
  I’ve never done anything, multimedia programming-wise, concerning subtitles. I guess all the entertainment I care about has always been in my native tongue. What a good excuse to program outside of my comfort zone!

  First, I needed to know how to access FFmpeg’s subtitling facilities. Fortunately, The Translator did the legwork on this matter so I didn’t have to figure it out.

  However, I intuitively had misgivings about this phase. I had heard that the subtitling process performs anti-aliasing. That means that the image would need to be promoted to a higher colorspace for this phase and that the anti-aliasing process would likely push the color count way past 256. Some quick tests revealed this to be the case, as the running color count would leap by several hundred colors as soon as the palette accounting algorithm encountered a subtitle.

  So I dug into the subtitle subsystem. I discovered that the subtitle library operates by creating a linked list of subtitle bitmaps that the client app must render. The bitmaps are comprised of 8-bit alpha transparency values that must be composited onto the target frame (i.e., 0 = transparent, 255 = 100% opaque). For example, the letter ‘H’:

                                    (with 00s removed)
  13 F8 41 00 00 00 00 68 E4  |  13 F8 41             68 E4    
  14 FF 44 00 00 00 00 6C EC  |  14 FF 44             6C EC
  14 FF 44 00 00 00 00 6C EC  |  14 FF 44             6C EC
  14 FF 44 00 00 00 00 6C EC  |  14 FF 44             6C EC
  14 FF DC D0 D0 D0 D0 E4 EC  |  14 FF DC D0 D0 D0 D0 E4 EC
  14 FF 7E 50 50 50 50 9A EC  |  14 FF 7E 50 50 50 50 9A EC
  14 FF 44 00 00 00 00 6C EC  |  14 FF 44             6C EC
  14 FF 44 00 00 00 00 6C EC  |  14 FF 44             6C EC
  14 FF 44 00 00 00 00 6C EC  |  14 FF 44             6C EC
  11 E0 3B 00 00 00 00 5E CE  |  11 E0 3B             5E CE
  

  To get around the color explosion problem, I chose a threshold value and quantized values above and below to 255 and 0, respectively. Further, the process chooses an appropriate color from the existing palette rather than introducing any new colors.

  Muxing Matters
  In order to force VMD into a general purpose media framework, a lot of special information needs to be passed around. Like many paletted codecs, the palette needs to be transmitted from the file demuxer to the video decoder via some side channel. For re-encoding, this also implies that the palette needs to make the trip from the video encoder to the file muxer. As if this wasn’t enough, individual VMD frames have even more data that needs to be ferried between the muxer and codec levels, including frame change boundaries. FFmpeg provides methods to do these things, but I could not always rely on the systems to relay the data in all cases. I was probably doing something wrong; I accept that. Instead, I just packed all the information at the front of an encoded frame and split it apart in the muxer.

  I could not quite figure out how to get the audio and video muxed correctly. As a result, neither FFmpeg nor the Phantasmagoria engine could replay the files correctly.

  Plan B
  Since I was having so much trouble creating an entirely new VMD file, likely due to numerous unknown bits of the file format, I thought of another angle: re-use the existing VMD file. For this approach, I kept the video encoder and file muxer that I created in the initial phase, but modified the file muxer to emit a special intermediate file. Then, I created a Python tool to repackage the original VMD file using compressed video data in the intermediate file.

  For this phase, I also implemented a command line switch for FFmpeg to disable subtitle blending, to make the feature feel like less of an unofficial hack, as though this nonsense would ever have a chance of being incorporated upstream.

  At this point, I was seeing some success with the complete, albeit roundabout, subtitling process. I constructed a subtitle file using “Spanish I Learned From Mexican Telenovelas” and the frames turned out fairly readable:

  
  

  

  “she cheated on him”

  
  

  

  “he’s a scumbag” … these random subtitles could fit surprisingly well!

  
  

  The few files that I tested appeared to work fine. But then I handed off my work to The Translator and he immediately found a bunch of problems. According to my notes, the problems mostly took the form of flashing, solid color frames. Further, I found tiny, mostly imperceptible flaws in my RLE compressor, usually only detectable by running strict comparison tools; but I wasn’t satisfied.

  At this point, I think I attempted to just encode the entire palette at the front of each frame, as allowed by the format, but that did not seem to fix any problems. My notes are not completely clear on this matter (likely because I was still trying to figure out the exact problem), but I think it had to do with FFmpeg inserting extra video frames in order to even out gaps in the video framerate.

  Sigh, Plan C
  At this point, I was getting tired of trying to force FFmpeg to do this. So I decided to minimize its involvement using lessons learned up to this point.

  The next pitch:

  1. Create a new C program that can open an existing VMD file and output an identical VMD file. I know this sounds easy, but the specific method of copying entails interpreting individual parts of the file and writing those individual parts to the new file. This is in preparation for…
  2. Import the VMD video decoder functions directly into the program to decode the individual video frames and re-encode them, replacing the video frames as the file is rewritten.
  3. Wire up the subtitle system. During the adventure to disable subtitle blending, I accidentally learned enough about interfacing to the subtitle library to just invoke it directly.
  4. Rewrite the RLE method so that it is 100% correct.

  Off to work I went. That part about lifting the existing VMD decoder functions out of their libavcodec nest turned out to not be that straightforward. As an alternative, I modified the decoder to dump the raw frames to an intermediate file. In doing so, I think I was able to avoid the issue of the duplicated frames that plagued the previous efforts.

  Also, remember how I was really pleased with the palette conversion technique in which I was able to leverage computer science big-O theory? By this stage, I had no reason to convert the paletted video to RGB in the first place; all of the decoding, subtitling and re-encoding operates in the paletted colorspace.

  This approach seemed to work pretty well. The final program is subtitle-vmd.c. The process is still a little weird. The modifications in my own FFmpeg fork are necessary to create an intermediate file that the new C tool can operate with.

  Next Steps
  The Translator has found some assorted bugs and corner cases that still need to be ironed out. Further, for extra credit, I need find the change windows for each frame to improve compression just a little more. I don’t think I will be trying for LZ compression, though.

  However, almost as soon as I had this whole system working, The Translator informed me that there is another, different movie format in play in the Phantasmagoria engine called ROBOT, with an extension of RBT. Fortunately, enough of the algorithms have been reverse engineered and re-implemented in ScummVM that I was able to sort out enough details for another subtitling project. That will be the subject of a future post.

  See Also:

  • Subtitling Sierra RBT Files: The followup in which I discuss how to scribble text on the other animation format

  The post Subtitling Sierra VMD Files first appeared on Breaking Eggs And Making Omelettes.

• Decoding an mp3 file using FFmpeg but sound is glitchy

  28 April 2017, by satyres

  After successfuly compiling the latest version of FFmpeg library and generated .a library in Ubuntu I’ve been struggling now for more than a week to play a simple mp3 file in Android without a success !
  The sound on my S4 working but it’s glitchy and stuttering
  I’ve followed this tutorial given by FFmpeg team in Github i’ve tried to use it in Android but no luck !
  here is the Native code.

   void Java_com_example_home_hellondk_MainActivity_audio_1decode_1example(JNIEnv * env, jobject obj, jstring file, jbyteArray array) {
  
          jboolean isfilenameCopy;
  
           const char * filename = (*env)->GetStringUTFChars(env, file,
  
                   &isfilenameCopy);
  
  jclass cls = (*env)->GetObjectClass(env, obj);
  
           jmethodID play = (*env)->GetMethodID(env, cls, "playSound", "([BI)V");
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int len;
  
      FILE *f, *outfile;
  
      uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
  
      AVPacket avpkt;
  
      AVFrame *decoded_frame = NULL;
  
  AVFormatContext* container=NULL;
  
      av_init_packet(&avpkt);
  
  
  
      printf("Decode audio file %s \n", filename);
  
  LOGE("Decode audio file %s\n", filename);
  
      /* find the MPEG audio decoder */
  
    /*  codec = avcodec_find_decoder(AV_CODEC_ID_MP3);
  
      if (!codec) {
  
          fprintf(stderr, "Codec not found\n");
  
          LOGE("Codec not found\n");
  
          exit(1);
  
      }*/
  
  int lError;
  
           if ((lError = avformat_open_input(&container, filename, NULL, NULL))
  
                   != 0) {
  
               LOGE("Error open source file: %d", lError);
  
               exit(1);
  
           }
  
           if ((lError =  avformat_find_stream_info(container,NULL)) < 0) {
  
               LOGE("Error find stream information: %d", lError);
  
               exit(1);
  
           }
  
           LOGE("Stage 1.5");
  
          LOGE("audio format: %s", container->iformat->name);
  
         LOGE("audio bitrate: %llu", container->bit_rate);
  
  
  
      int stream_id = -1;
  
          // To find the first audio stream. This process may not be necessary
  
          // if you can gurarantee that the container contains only the desired
  
          // audio stream
  
             LOGE("nb_streams: %d", container->nb_streams);
  
          int i;
  
          for (i = 0; i < container->nb_streams; i++) {
  
              if (container->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  
                  stream_id = i;
  
                   LOGE("stream_id: %d", stream_id);
  
                  break;
  
              }
  
          }
  
  
  
          AVCodecContext* codec_context = container->streams[stream_id]->codec;
  
          codec = avcodec_find_decoder(codec_context->codec_id);
  
           LOGE("stream_id: %d", stream_id);
  
          LOGE("codec %s", codec->name);
  
          if (!codec) {
  
              fprintf(stderr, "codec not found\n");
  
              exit(1);
  
          }
  
  
  
      c = avcodec_alloc_context3(codec);
  
      if (!c) {
  
          fprintf(stderr, "Could not allocate audio codec context\n");
  
            LOGE("Could not allocate audio codec context\n");
  
          exit(1);
  
      }
  
  
  
      /* open it */
  
      if (avcodec_open2(c, codec, NULL) < 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
           LOGE("Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      f = fopen(filename, "rb");
  
      if (!f) {
  
          fprintf(stderr, "Could not open %s\n", filename);
  
          LOGE("Could not open %s\n",filename);
  
          exit(1);
  
      }
  
  
  
  
  
  
  
          avpkt.data = inbuf;
  
          avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
  
          LOGE("Stage 5");
  
      /* decode until eof */
  
  
  
      while (1) {
  
  
  
      if ((len = av_read_frame(container, &avpkt)) < 0)
  
                         break;
  
  
  
          if (avpkt.stream_index == stream_id)
  
                       {
  
        if (!decoded_frame) {
  
                                               if (!(decoded_frame = av_frame_alloc())) {
  
                                                   fprintf(stderr, "Could not allocate audio frame\n");
  
                                                    LOGE("Could not allocate audio frame\n");
  
                                                   exit(1);
  
                                               }
  
                                           }
  
                              int got_frame = 0;
  
                                len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
  
                               LOGE("len=%d",len);
  
                               if (len < 0)
  
                               {
  
                                   LOGE("Error decoding audio\n");
  
                                   continue;
  
                               }
  
  
  
                               if (got_frame)
  
                               {
  
                                    LOGE("begin frame decode\n");
  
                                    int data_size = av_samples_get_buffer_size(NULL, c->channels,decoded_frame->nb_samples,c->sample_fmt, 1);
  
                                  if (data_size>0)
  
                                  {
  
                                    LOGE("after frame decode %d\n",data_size);
  
  
  
                                   jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
  
                                    memcpy(bytes, decoded_frame->data[0], data_size);
  
                                    (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
                                    (*env)->CallVoidMethod(env, obj, play, array, data_size);
  
  
  
                                   }
  
                                   else
  
                                   {
  
                                   fprintf(stderr, "Failed to calculate data size\n");
  
                                    exit(1);
  
  
  
                                   }
  
  
  
                               }
  
  
  
                               avpkt.size -= len;
  
                               avpkt.data += len;
  
                               avpkt.pts = AV_NOPTS_VALUE;
  
                              if (avpkt.size < AUDIO_REFILL_THRESH)
  
                              {
  
                                  memmove(inbuf, avpkt.data, avpkt.size);
  
                                  avpkt.data = inbuf;
  
                                  len = fread(avpkt.data + avpkt.size, 1, AUDIO_INBUF_SIZE - avpkt.size, f);
  
                                  if (len > 0)
  
                                      avpkt.size += len;
  
                              }
  
  
  
                         }
  
  
  
  
  
  
  
  
  
      }
  
  
  
  
  
      fclose(f);
  
  
  
      avcodec_free_context(&c);
  
  av_frame_free(&decoded_frame);
  
      }

  The Java code :

  package com.example.home.hellondk;
  
  
  
  import android.media.AudioFormat;
  
  import android.media.AudioManager;
  
  import android.media.AudioTrack;
  
  import android.media.MediaPlayer;
  
  import android.support.v7.app.AppCompatActivity;
  
  import android.os.Bundle;
  
  import android.util.Log;
  
  
  
  import java.io.File;
  
  import java.io.FileNotFoundException;
  
  import java.io.FileOutputStream;
  
  import java.io.IOException;
  
  
  
  public class MainActivity extends AppCompatActivity {
  
      static {
  
          System.loadLibrary("MyLibraryPlayer");
  
      }
  
      public native void createEngine();
  
  
  
      public native void audio_decode_example(String outfilename, byte[] array);
  
  
  
  
  
  
  
      private AudioTrack track;
  
      private FileOutputStream os;
  
  
  
      @Override
  
      protected void onCreate(Bundle savedInstanceState) {
  
          super.onCreate(savedInstanceState);
  
          setContentView(R.layout.activity_main);
  
          createEngine();
  
  
  
   /*        MediaPlayer mp = new MediaPlayer();
  
          mp.start();*/
  
  
  
          int bufSize = AudioTrack.getMinBufferSize(32000,
  
                  AudioFormat.CHANNEL_CONFIGURATION_STEREO,
  
                  AudioFormat.ENCODING_PCM_16BIT);
  
  
  
          track = new AudioTrack(AudioManager.STREAM_MUSIC,
  
                  32000,
  
                  AudioFormat.CHANNEL_CONFIGURATION_STEREO,
  
                  AudioFormat.ENCODING_PCM_16BIT,
  
                  bufSize,
  
                  AudioTrack.MODE_STREAM);
  
  
  
          byte[] bytes = new byte[bufSize];
  
  
  
  
  
     audio_decode_example("/storage/emulated/0/test.mp3", bytes);
  
  
  
      }
  
  
  
      void playSound(byte[] buf, int size) {
  
          //android.util.Log.v("ROHAUPT", "RAH Playing");
  
          if (track.getPlayState() != AudioTrack.PLAYSTATE_PLAYING)
  
              track.play();
  
          track.write(buf, 0, size);
  
  
  
  
  
      }
  
  }

  Thank you so much for your help.
  Kind regards

• Playing Mp3 file using FFmpeg on Android

  2 April 2017, by satyres

  This question has been asked a lot but no code has worked for me . i’ve been able to play a file decoded with ffmpeg on Android but it’s noisy and glitchy.
  i’ve found code in book called "linux sound programming" using latest ffmpeg version to decode an mp3 file.
  the code tries to decode an mp3 file to pcm and then put it in a file called output.
  what i want to do is to get the byte decoded on the fly and send them to AudioTrack in Java.

   void JNICALL Java_com_example_home_hellondk_MainActivity_loadFile
  
          (JNIEnv* env, jobject obj,jstring file,jbyteArray array)
  
  {
  
  
  
   jboolean isfilenameCopy;
  
           const char * filename = (*env)->GetStringUTFChars(env, file,
  
                   &isfilenameCopy);
  
  jclass cls = (*env)->GetObjectClass(env, obj);
  
           jmethodID play = (*env)->GetMethodID(env, cls, "playSound", "([BI)V");
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int len;
  
      FILE *f, *outfile;
  
      uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
  
      AVPacket avpkt;
  
      AVFrame *decoded_frame = NULL;
  
  AVFormatContext* container=NULL;
  
      av_init_packet(&avpkt);
  
   int num_streams = 0;
  
      int sample_size = 0;
  
      printf("Decode audio file %s \n", filename);
  
  LOGE("Decode audio file %s\n", filename);
  
      /* find the MPEG audio decoder */
  
    /*  codec = avcodec_find_decoder(AV_CODEC_ID_MP3);
  
      if (!codec) {
  
          fprintf(stderr, "Codec not found\n");
  
          LOGE("Codec not found\n");
  
          exit(1);
  
      }*/
  
  int lError;
  
           if ((lError = avformat_open_input(&container, filename, NULL, NULL))
  
                   != 0) {
  
               LOGE("Error open source file: %d", lError);
  
               exit(1);
  
           }
  
           if ((lError =  avformat_find_stream_info(container,NULL)) < 0) {
  
               LOGE("Error find stream information: %d", lError);
  
               exit(1);
  
           }
  
           LOGE("Stage 1.5");
  
          LOGE("audio format: %s", container->iformat->name);
  
         LOGE("audio bitrate: %llu", container->bit_rate);
  
  
  
      int stream_id = -1;
  
          // To find the first audio stream. This process may not be necessary
  
          // if you can gurarantee that the container contains only the desired
  
          // audio stream
  
             LOGE("nb_streams: %d", container->nb_streams);
  
          int i;
  
          for (i = 0; i < container->nb_streams; i++) {
  
              if (container->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  
                  stream_id = i;
  
                   LOGE("stream_id: %d", stream_id);
  
                  break;
  
              }
  
          }
  
  
  
          AVCodecContext* codec_context = container->streams[stream_id]->codec;
  
          codec = avcodec_find_decoder(codec_context->codec_id);
  
           LOGE("stream_id: %d", stream_id);
  
          LOGE("codec %s", codec->name);
  
          if (!codec) {
  
              fprintf(stderr, "codec not found\n");
  
              exit(1);
  
          }
  
  
  
      c = avcodec_alloc_context3(codec);
  
      if (!c) {
  
          fprintf(stderr, "Could not allocate audio codec context\n");
  
            LOGE("Could not allocate audio codec context\n");
  
          exit(1);
  
      }
  
  
  
      /* open it */
  
      if (avcodec_open2(c, codec, NULL) < 0) {
  
          fprintf(stderr, "Could not open codec\n");
  
           LOGE("Could not open codec\n");
  
          exit(1);
  
      }
  
  
  
      f = fopen(filename, "rb");
  
      if (!f) {
  
          fprintf(stderr, "Could not open %s\n", filename);
  
          LOGE("Could not open %s\n",filename);
  
          exit(1);
  
      }
  
      const char *outfilename;
  
   outfile = fopen(outfilename, "wb");
  
      if (!outfile) {
  
          av_free(c);
  
          exit(1);
  
      }
  
  
  
  
  
          avpkt.data = inbuf;
  
          avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
  
          LOGE("Stage 5");
  
      /* decode until eof */
  
  
  
       while (avpkt.size > 0) {
  
             int got_frame = 0;
  
             if (!decoded_frame) {
  
                 if (!(decoded_frame = av_frame_alloc())) {
  
                     fprintf(stderr, "out of memory\n");
  
                     exit(1);
  
                 }
  
                 } else {
  
                             av_frame_unref(decoded_frame);
  
                         }
  
                         printf("Stream idx %d\n", avpkt.stream_index);
  
                         len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
  
                         if (len < 0) {
  
                             fprintf(stderr, "Error while decoding\n");
  
                             exit(1);
  
                         }
  
                         if (got_frame) {
  
                             printf("Decoded frame nb_samples %d, format %d\n",
  
                                    decoded_frame->nb_samples,
  
                                    decoded_frame->format);
  
                             if (decoded_frame->data[1] != NULL)
  
                                 printf("Data[1] not null\n");
  
                             else
  
                                 printf("Data[1] is null\n");
  
                             /* if a frame has been decoded, output it */
  
                             int data_size = av_samples_get_buffer_size(NULL, c->channels,
  
                                                                        decoded_frame->nb_samples,
  
                                                                        c->sample_fmt, 1);
  
                             // first time: count the number of  planar streams
  
                             if (num_streams == 0) {
  
                                 while (num_streams < AV_NUM_DATA_POINTERS &&
  
                                        decoded_frame->data[num_streams] != NULL)
  
                                     num_streams++;
  
                                 printf("Number of streams %d\n", num_streams);
  
                             }
  
                             // first time: set sample_size from 0 to e.g 2 for 16-bit data
  
                             if (sample_size == 0) {
  
                                 sample_size =
  
                                     data_size / (num_streams * decoded_frame->nb_samples);
  
                             }
  
                             int m, n;
  
  
  
                             for (n = 0; n < decoded_frame->nb_samples; n++) {
  
                                 // interleave the samples from the planar streams
  
                                 for (m = 0; m < num_streams; m++) {
  
                                    fwrite(&decoded_frame->data[m][n*sample_size],
  
                                            1, sample_size, outfile);
  
  
  
                                 }
  
                             }
  
  
  
                  /*            jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
  
                              memcpy(bytes, decoded_frame->data[1], data_size);
  
                              (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
                             (*env)->CallVoidMethod(env, obj, play, array, data_size);
  
  
  
  */
  
                         }
  
                         avpkt.size -= len;
  
                         avpkt.data += len;
  
                         if (avpkt.size < AUDIO_REFILL_THRESH) {
  
                             /* Refill the input buffer, to avoid trying to decode
  
                              * incomplete frames. Instead of this, one could also use
  
                              * a parser, or use a proper container format through
  
                                * libavformat. */
  
                                          memmove(inbuf, avpkt.data, avpkt.size);
  
                                          avpkt.data = inbuf;
  
                                          len = fread(avpkt.data + avpkt.size, 1,
  
                                                      AUDIO_INBUF_SIZE - avpkt.size, f);
  
                                          if (len > 0)
  
                                              avpkt.size += len;
  
                                      }
  
                                  }
  
  
  
      fclose(f);
  
  
  
      avcodec_free_context(&c);
  
  av_frame_free(&decoded_frame);
  
  
  
  }

  the decoded bytes are in this section
  fwrite(&decoded_frame->data[m][n*sample_size],                                           1, sample_size, outfile);

  the code that let you send bytes to java is this :

    jbyte *bytes = (*env)->GetByteArrayElements(env, array, NULL);
  
    memcpy(bytes, decoded_frame->data[0], data_size);
  
   (*env)->ReleaseByteArrayElements(env, array, bytes, 0);
  
   (*env)->CallVoidMethod(env, obj, play, array, data_size);

  i’ve been working on it now for more than a week and nothing worked for me.

  Thank you in advance for your help